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DSP Answers PDF
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OBJECTIVES
To study DFT and its computation
To study the design techniques for digital filters
To study the finite word length effects in signal processing
To study the non-parametric methods of power spectrum estimations
To study the fundamentals of digital signal processors.
UNIT I FFT 9
Introduction to DFT Efficient computation of DFT Properties of DFT FFT algorithms
Radix-2 FFT algorithms Decimation in Time Decimation in Frequency algorithms Use of
FFT algorithms in Linear Filtering and correlation.
1. John G Proakis, Dimtris G Manolakis, Digital Signal Processing Principles, Algorithms and
Application, PHI, 3rd Edition, 2000,
2. B.Venkataramani & M. Bhaskar, Digital Signal Processor Architecture, Programming and
Application, TMH 2002. (UNIT V)
REFERENCES
1. Alan V Oppenheim, Ronald W Schafer, John R Back, Discrete Time Signal Processing, PHI, 2nd
Edition 2000,
2. Avtar singh, S.Srinivasan DSP Implementation using DSP microprocessor with Examples from
TMS32C54XX -Thamson / Brooks cole Publishers, 2003
3. S.Salivahanan, A.Vallavaraj, Gnanapriya, Digital Signal Processing, McGraw-Hill / TMH, 2000
4. Johny R.Johnson :Introduction to Digital Signal Processing, Prentice Hall, 1984.
5. S.K.Mitra, Digital Signal Processing- A Computer based approach, Tata McGraw-Hill, 1998,
New Delhi.
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EC2302 DIGITAL SIGNAL PROCESSING 23
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EC2302 DIGITAL SIGNAL PROCESSING 51
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DEPARTMENT OF ECE, ADHIPARASAKTHI COLLEGE OF ENGINEERING, KALAVAI.
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EC2302 DIGITAL SIGNAL PROCESSING 53
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DEPARTMENT OF ECE, ADHIPARASAKTHI COLLEGE OF ENGINEERING, KALAVAI.
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EC2302 DIGITAL SIGNAL PROCESSING 55
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DEPARTMENT OF ECE, ADHIPARASAKTHI COLLEGE OF ENGINEERING, KALAVAI.
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EC2302 DIGITAL SIGNAL PROCESSING 57
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DEPARTMENT OF ECE, ADHIPARASAKTHI COLLEGE OF ENGINEERING, KALAVAI.
1. What is DFT?
It is a finite duration discrete frequency sequence, which is obtained by sampling one period of
Fourier transform. Sampling is done at N equally spaced points over the period extending from
w=0 to 2.
Here k=0,1,2N-1
N=L+M-1
12. What is the way to reduce number of arithmetic operations during DFT computation?
Number of arithmetic operations involved in the computation of DFT is greatly reduced by using
different FFT algorithms as follows.
If X3 (K) = X1 (K) X2 (K), then the sequence x3 (n) can be obtained by circular convolution, defined
as
18. What is the speed of improvement factor in calculating 64-point DFT of a sequence using
direct computation and computation and FFT algorithms?
Or
Calculate the number of multiplications needed in the calculation of DFT and FFT with 64-
point sequence.
N2=642=4096.
20. Calculate the number of multiplications needed in the calculation of DFT using FFT algorithm
with using FFT algorithm with 32-point sequence.
For N-point DFT the number of complex multiplications needed using FFT algorithm is
N/2 log2N.
22. How many multiplications and additions are required to compute N-point DFT using redix-2
FFT?
The number of multiplications and additions required to compute N-point DFT using redix-2 FFT are
N log2N and N/2 log2N respectively.
25. What are the differences and similarities between DIF and DIT algorithms?
Differences:
1. For DIT, the input is bit reversal while the output is in natural order, whereas for DIF, the
input is in natural order while the output is bit reversed.
2. The DIF butterfly is slightly different from the DIT butterfly, the difference being that the
complex multiplication takes place after the add-subtract operation in DIF.
Similarities:
Both algorithms require same number of operations to compute the DFT. Both algorithms
can be done in place and both need to perform bit reversal at some place during the computation.
The right hand side of the above equation is DFT of the sequence X*(K) and may be computed using
any FFT algorithm. The desired output sequence x (n) can be then obtained by complex conjugating
the DFT of the above equation and dividing by N to give
28. Draw the 4-point radix-2 DIT-FFT butterfly structure for DFT.
29. Draw the 4-point radix-2 DIF-FFT butterfly structure for DFT.
30. Draw the basic butterfly diagram for DIT, DIF algorithm.
1 Gives the harmonic content of a periodic Gives the frequency information for an
time function. aperiodic signal.
Impulse invariance
Bilinear transformation.
3) Which of the methods do you prefer for designing IIR filters? Why?
Bilinear transformation is best method to design IIR filter, since there is no aliasing in it.
5) What is prewarping?
Prewarping is the method of introducing nonlinearly in frequency relationship to
compensate warping effect.
8) Where left hand side and right hand side are mapped in z-plane in bilinear transformation?
Left hand side -- Inside unit circle
12) What is the advantage of designing IIR Filters using pole-zero plots?
The frequency response can be located exactly with the help of poles and zeros.
14) Write equations for transfer functions of Chebyshev filter and Butterworth filter.
15) What are the equations of lowpass to lowpass & lowpass to highpass transformation?
16) Pole-zero plot of the filter is given in fig.1. Determine its difference equation.
ii) It is governed by linear difference equation. ii) It is governed by linear difference equation.
25) Obtain the impulse response of digital filter to correspond to an analog filter with impulse
response ha(t) = 0.5 e-2t and with a sampling rate of 1.0kHz using impulse invariant method.
26) How analog poles are mapped to digital poles in impulse invariant transformation?
In impulse invariant transformation the mapping of analog to digital poles are as follows,
The analog poles on the left half of s-plane are mapped into the interior of unit circle in
z-plane.
The analog poles on the imaginary axis of s-plane are mapped into the unit circle in the
z-plane.
The analog poles on the right half of s-plane are mapped into the exterior of unit circle
in z-plane.
29) Given that, Ha(s) = 1 / (s +1). By impulse invariant method, obtain the digital transfer function
and the difference equation of digital filter.
30) Write the impulse invariant transformation used to transform real poles with and without
multiplicity.
31) Write the impulse invariant transformation used to transform complex conjugate poles.
34) What is the relationship between digital and analog frequency in Bilinear transformation?
Bilinear transformation the digital frequency is given by,
Digital frequency, w =
36) How the analog frequency is mapped to digital frequency in bilinear transformation?
37) How the order of the filter affects the frequency response of Butterworth filter.
The magnitude response of butterworth filter is shown in figure, from which it can be observed
that the magnitude response approaches the ideal response as the order of the filter is increased.
39) How will you choose the order N for a butterworth filter.
ii. The poles lie on a circle in s-plane. ii. The poles lie on a ellipse in s-plane.
iii. The magnitude response is maximally flat iii. The magnitude response is equiripple in
at the origin and monotonically decreasing passband and monotonically decreasing in the
function of . stopband.
iv. The normalized magnitude response has a iv. The normalized magnitude response has a
value of 1 / 2 at the cutoff frequency c. value of 1 / (1+ 2) at the cutoff frequency c.
The IIR filters are of recursive type, whereby the present output sample depends on the
present input, past input samples and output samples.
The FIR filters are of non recursive type, whereby the present output sample depends on the
present input, and previous output samples.
1. These filters can be easily designed to These filters do not have linear phase.
have perfectly linear phase.
The phase distortion is introduced when the phase characteristics of a filter is not linear
within the desired frequency band.
The delay distortion is introduced when the delay is not constant within the desired
frequency range.
13. What is the necessary and sufficient condition for the linear phase characteristic of an FIR
filter?
The necessary and sufficient condition for the linear phase characteristic of an FIR filter is that the phase function should be a linear
function of w, which in turn requires constant phase and group delay.
14. What are the conditions to be satisfied for constant phase delay in linear phase FIR filters?
The conditions for constant phase delay ARE
15. How constant group delay & phase delay is achieved in linear phase FIR filters?
The following conditions have to be satisfied to achieve constant group delay & phase delay.
16. What are the possible types of impulse response for linear phase FIR filters?
There are four types of impulse response for linear phase FIR filters
17. List the well-known design techniques of linear phase FIR filters.
There are three well-known design techniques of linear phase FIR filters. They are
20. Draw the direct form realization of a linear phase FIR system for N is odd.
21. Draw the direct form realization of a linear phase FIR system for N is even.
23. What are the desirable characteristics of the frequency response of window function?
The desirable characteristics of the frequency response of window function are
The width of the mainlobe should be small and it should contain as much of the total
energy as possible.
The sidelobes should decrease in energy rapidly as w tends to .
24. Write the procedure for designing FIR filter using frequency-sampling method.
Choose the desired (ideal) frequency response Hd(w).
Take N-samples of Hd(w) to generate the sequence
Take inverse DFT of to get the impulse response h(n).
The transfer function H(z) of the filter is obtained by taking z-transform of impulse
response.
25. What are the drawback in FIR filter design using windows and frequency sampling method?
How it is overcome?
The FIR filter design using windows and frequency sampling method does not have
This drawback can be overcome by designing FIR filter using Chebyshev approximation
technique. In this technique an error function is used to approximate the ideal frequency
response, in order to satisfy the desired specifications.
26. Write the expression for frequency response of rectangular window and sketch the magnitude
response.
28. List the features of FIR filter designed using rectangular window.
The width of the transition region is related to the width of the mainlobe of window
spectrum.
Gibbs oscillations are noticed in the passband and stopband.
The attenuation in the stopband is constant and cannot be varied.
29. Why Gibbs oscillations are developed in rectangular window and how it can be eliminated or
reduced?
The Gibbs oscillations in rectangular window are due to the sharp transitions from 1 to 0 at
the edges of window sequence.
i) The width of main lobe in window i) The width of main lobe in window
spectrum is 4/N spectrum is 8/N
ii) The maximum side lobe magnitude in ii) The maximum side lobe magnitude in
window spectrum is 13dB. window spectrum is 31dB.
iii) In window spectrum the side lobe iii) In window spectrum the side lobe
magnitude slightly decreases with increasing magnitude decreases with increasing w.
w.
iv) In FIR filter designed using hamming
iv) In FIR filter designed using rectangular window the minimum stop band attenuation
window the minimum stop band attenuation is 44dB.
is 22dB.
iii) In window spectrum the sidelobe iii) In window spectrum the sidelobe
magnitude slightly decreases with increasing magnitude remains constant.
w.
iv) In FIR filter designed using hamming
iv) In FIR filter designed using rectangular window the minimum stopband attenuation
window the minimum stopband attenuation is 44dB.
is 22dB.
38. What is the mathematical problem involved in the design of window function?
The mathematical problem involved in the design of window function(or sequence) is
that of finding a time-limited function whose Fourier Transform best approximates a band
limited function. The approximation should be such that the maximum energy is confined to
mainlobe for a given peak sidelobe amplitude.
The width of the mainlobe in the window spectrum can be varied by varying the length
N of the window sequence.
UNIT III
FINITE WORD LENGTH EFFECTS
In all the three formats, the positive number is same but they differ only in
representing negative numbers.
7. What is truncation?
The truncation is the process of reducing the size of binary number by discarding all bits
less significant than the least significant bit that is retained. In truncation of a binary
number of b bits all the less significant bits beyond bth bit are discarded.
8. What is rounding?
Rounding is the process of reducing the size of a binary number to finite word size of b-bits
such that, the rounded b-bit number is closest to the original unquantized number.
The A/D process generates two types of errors. They are quantization error and
saturation error. The quantization error is due to representation of the sampled
signal by a fixed number of digital levels. The saturation errors occur when the
analog signal exceeds the dynamic range of A/D converter.
13. What is steady state output noise power due to input quantization?
The input signal to digital system can be considered as a sum of unquantized signal
and error signal due to input quantization. The response of the system can be
expressed as a summation of response due to unquantized input and error signal.
The response of the system due to error signal is given by convolution of error
signal and impulse response. The variance of response of the system for error
signal is called state output noise power.
20. What are the assumptions made regarding the statistical independence of the
various noise sources in the digital filter?
The assumptions made regarding the statistical independence of the noise sources
are,
1. Any two different samples from the same noise source are uncorrelated.
2. Any two different noise source, when considered, as random processes are
uncorrelated.
3. Each noise source is uncorrelated with the input sequence.
25. How the system output cam be brought out of limit cycles?
The system output can be brought out of limit cycle by applying an input of large
magnitude, which is sufficient to drive the system out of limit cycle.
UNIT V
(i) Hardware designed for efficient execution of specific DSP algorithms such as
digital filter, FFT.
(ii) Hardware designed for specific applications, for example telecommunication,
digital audio.
1. Instruction fetch
2. Instruction decode
3. Instruction execute
Method.
For getting down these operations we need the help of adders and multipliers. The
combination of these accumulator and multiplier is called as multiplier accumulator.
(i) Fetch
(ii) Decode
(iii) Read
(iv) Execution
8. In a non-pipeline machine, the instruction fetch, decode and execute take 30 ns, 45 ns
and 25 ns respectively. Determine the increase in throughput if the instruction were pipelined.
Assume a 5ns pipeline overhead in each stage and ignore other delays.
= 100/135 = 0.7407
But in the case of pipeline machine, the clock speed is determined by the speed of the slowest stage
plus overheads.
In our case is = 45 ns + 5 ns =50 ns
9.Assume a memory access time of 150 ns, multiplication time of 100 ns, addition time of 100
ns and overhead of 10 ns at each pipe stage. Determine the throughput of MAC
Direct addressing.
Indirect addressing.
Bit-reversed addressing.
Immediate addressing.
11.What are the instructions used for block transfer in C5X Processors?
The BLDD, BLDP and BLPD instructions use the BMAR to point at the source or
destination space of a block move. The MADD and MADS also use the BMAR to address
an operand in program memory for a multiply accumulator operation
The advantage of this addressing mode is that the address of the block of
memory to be acted upon can be changed during execution of the program.
16. Write short notes about arithmetic logic unit and accumulator.
The 32-bit general-purpose ALU and ACC implement a wide range of arithmetic and
logical functions, the majority of which execute in a single clock cycle. Once an
operation is performed in the ALU, the result is transferred to the ACC, where additional
operations, such as shifting, can occur. Data that is input to the ALU can be scaled by
the prescaler.
storage. Indirect auxiliary register addressing allows placement of the data memory
address of an instruction operand into one of the AR. The ARs are pointed to by a 3-bit
auxiliary register pointer (ARP) that is loaded with a value from 0-7, designating AR0-
AR7, respectively.